[Freeswitch-users] Some problems with bypass-media

[Neven Boric]

Hi,

I have been using FS for a long time on a local server without issues, but
now I want to move to a remote server to support some new usage scenarios.
I'm trying to use inbound-bypass-media=true to keep the audio out of the
server. This mostly works, but I have two different issues, depending on
which FS version I use:

- I first tried with the 1.2 branch, as that was the version I was using
locally. The issue with that version is that when a phone holds and then
unholds a call, I get no audio on the phone that started the hold. I found
option bypass-media-after-hold, but realized that it was not included in
the 1.2 branch, so I "ported" commits 3ecb504, 8fa385b and e4e9b1b9f93 from
the master branch. Now FS correctly tries to restore direct media between
the endpoints with reINVITEs, but unfortunately includes a 'sendonly' in
the last SDP to the phone that unholds, so I still get no audio. I haven't
found a way to fix this.

- I also tried with the 1.4 branch and hold/unhold works correctly, but now
attended transfer doesn't work, FS after some time ends the call on the
side that was put on hold waiting to be transferred. This time seems to be
inconsistent, and sometimes the transfer actually works, so maybe it is a
timing issue. This seems similar/related to FS-4038.

I can get logs and SIP captures, what would be the preferred way to provide
them (pcap, inline text, pastebin)? Or maybe it's better to go ahead and
file a bug?

Best regards
Neven Boric
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150708/118677b1/attachment.html